The telephony network has evolved over many decades; particularly since the 1960s, when stored-program-controlled switches with specific engineering requirements for voice telephony were deployed. Engineering requirements that are critical to achieve high-quality voice telephony include: (1) guaranteed bandwidth, (2) minimum network delay, and (3) a very low probability (≦0.1%) that a network would block an attempted call or drop an ongoing call. Transmissions that satisfy these engineering requirements are defined for the purposes of this specification as “high-quality traffic”.
The Internet, on the other hand, has evolved over the last decade to transport mostly computer data originating from web browsing, document and music downloading, and e-commerce. Yet traffic on the Internet surpassed the total traffic on the voice telephony network several years ago. While the volume of traffic on the Internet is increasing, network requirements for computer data traffic need not be as stringent as for voice telephony, however.
The operations and maintenance of the separate voice telephony network is expensive given that the ratio of telephony traffic to Internet data traffic will continue to decrease.
While Internet Protocol (IP) telephony, also known as voice over IP (VoIP) service, has been introduced several years ago, the quality of VoIP service has not yet been at par with the quality of service on the pre-existing voice telephony network. The Internet substantially operates in a “best-effort” delivery mode for packets that are traversing its network. Packets are forwarded from one node (i.e., a signal termination or transit point in a network, such as an IP router in an IP network or a cross-connect in a voice network) in the network to the next node with no regard for the capacity available in the link between the nodes or delay that the packets might encounter, and without a guarantee that the packets will be delivered to their final destination. Transmissions made in a best-effort manner, as described above, are defined for the purposes of this specification as “best-effort traffic”.
The quality of VoIP services is degraded by multiple factors that include: (a) latency for a packet to travel from source to destination (i.e. transit delay), (b) variation in latency from one packet to another commonly known as jitter, and (c) loss of packets. In an effort to mitigate the degradation in service quality due to these factors, some characteristics of the circuit-switched telephony network have been incorporated into the VoIP service network. For example, Multi-Protocol Label Switching (MPLS) has been developed and deployed in some best-effort packet switching networks in order to mimic the circuit-switching features of high-quality service. In spite of these improvements for VoIP implementation in IP networks, the basic nature of the packet switching devices, wherein the header of each packet is examined and then the packet is forwarded to a next destination (i.e., the next “hop”), remains a fundamental challenge to providing ubiquitous high-quality VoIP service over an IP network. On the other hand, costly maintenance and operation of the old circuit-switched voice telephony network for a diminishing fraction of the total information traffic offers a challenge as well as an opportunity to offer both best-effort traffic and high-quality traffic in an integrated network.
A typical telephony network comprises a set of stored program controlled (SPC) switching systems, known as Class V switches, which are interconnected by digital links, referred to as “trunk” lines. The Class V switches provide the interconnection between customer voice access lines and the trunk-lines. Each Class V switch can terminate up to tens of thousands of customer access voice lines. A circuit (i.e., the connection from a first customer access line to a second customer access line through the network) is established through one or more Class V switches, and the connectivity is based on a signaling scheme, such as Signaling System 7 (SS7).
Each voice channel is carried on a single 64 Kb/s digital stream known as a DS0 signal. Each trunk-line carries multiple DS0 signals in a higher level of digital stream; 24 DS0 signals compose a DS1 signal in North America, while 32 DS0 signals compose an E1 signal in Europe and elsewhere. The plurality of DS0 signals are multiplexed into the higher level signals using Time Division Multiplexing (TDM). Hence the digital voice telephony network is also called TDM network. The DS1 signals are further multiplexed to higher bit streams using an older asynchronous TDM scheme to DS2 (6.312 Mb/s) and DS3 (44.712 Mb/s) digital signals.
A newer TDM technique known as Synchronous Optical Network (SONET) mostly used in North America and Synchronous Digital Hierarchy (SDH) in other parts of the world are used to multiplex the lower level digital signals, such as DS1 and DS3, to a much higher-speed signal, such as OC-48 (2.5 Gb/s) or OC-192 (10 Gb/s), a number of which in turn are wave division multiplexed by a Wavelength Division Multiplexing (WDM) system for long distance transport over optical fiber. Even though SONET and SDH technologies are very similar now, historically they were developed for multiplexing 24-cahnnel and 32-channel digital streams to synchronous optical signals. However, the two hierarchies and standards evolved to converge so that there is practically no difference between the two. Thus henceforth only 24-channel terminology will be used even though all aspects of this application shall apply equally to the 32-channel system.
A transport network multiplexes low-level signals, transports the multiplexed high-speed signal over a long distance, and then demultiplexes it to extract the lower level signal for connection to Class V switches and other service access systems at the distant location. In addition to the management of the interconnectivity of the service access systems, a transport network provides another key network function, namely, network protection and restoration. In the case of a network link or equipment failure, the transport network finds alternate route to restore affected connections via alternate links or routes among the service access systems within a fraction of a second, thereby making the service systems immune to network failures.
While the transport network provides robust connectivity and management of the connectivity of voice-switches, such as Class V switches, it does not participate in making decisions on how a DS0 signal within a trunk-line is used to route a call. The voice-switches provide the routing function using a signaling scheme, such as SS7, and a different signaling scheme, such as tip-and-ring, between a voice-switch and a user terminal, such as a telephone.
The telephony network meets the key voice service requirements by its nature of being a circuit-switching network and by proper engineering of the network. A fixed bandwidth requirement to transmit a voice signal between two parties is automatically met by the nature of the circuit-switched network. When a connection is made it is of a fixed bandwidth typically 64 Kb/s between the two devices and the circuit is used exclusively for one call during the call session. Telephony network meets the minimum delay requirement also because of the nature of circuit-switching technology. A circuit switch device does not need to continuously process the circuit in the transit nodes once the connection is established. Thus, the delay for a voice signal from one end to the other is kept at a minimum, which is essentially the transit delay, required for the electrical or optical signal to travel over the transmission medium. The requirement on minimum call blocking probability is met by proper engineering. Using traffic characteristics such as incoming call volume, call duration etc. and traffic engineering techniques, an appropriate number of trunk-lines are configured in a switch to keep the call blocking probability to a required minimum level. In addition, call-dropping probability is kept at a minimum by using the network protection and restoration techniques in the transport network to further enhance the reliability of the trunk-lines.
An IP network is essentially a network of computers to exchange data usually between server and client computers. An IP network comprises a set of IP routers that are connected by high-speed SONET/SDH links on one side and the server and client computers on the other. The application programs running in the client/server computers generate information packets each with a destination address added at the header of the packet. The routers then forward each packet independently, using a set of routing protocols.
Unlike in a telephony network, there is no signaling to first establish an end-to-end fixed bandwidth connection for a packet stream in an IP network. Instead, each router has the means to generate and update a routing table, which includes all reachable IP addresses in the network. A router upon receiving a packet looks at the destination address in the packet; checks the routing table for best match with the destination address and decides which interface it will forward the packet to its next hop router. This process is repeated in each router on the route of a packet until it reaches the final destination.
The computer application program at the other end removes the added information bytes from the packets and then assembles the packets to generate the information for presentation to the user, whether it is a document, a picture or music. The packets from a single application may even arrive via different routes and in a different sequence than the one generated at the source. The application program stores these packets and rearranges the sequence for the information to be presented to the user. The process of storing the packets for rearrangement adds further delay to the information transport. In addition, the transmission links between routers are shared among different applications passing through a router. Thus, there is no guarantee that a specific application session will have a minimum bit rate capacity or bandwidth along its routes.
In an IP network, packets are discarded when there is a shortage in link capacity. When packets are discarded, the application layer program detects that packets are missing, and requests the sender of the missing packets to resend them. The process of resending also adds to the overall delay. Delay, however, is not detrimental to best-effort traffic. The IP network is not well suited, however, for applications wherein high-quality traffic, such as VoIP, video telephony and videoconferencing, is transmitted.
IP routers are being updated with some capabilities to deal with these impairments. For example, some applications may extend the headers of high-quality traffic packets with one or more markers that designate these packets for special handling. These markers may include: source-routing, in which the source specifies the entire route to be taken by the packets; a quality of service (QoS) indicator, which specifies routing priority for high-quality traffic packets over best-effort traffic packets; or routing protocols, such as MPLS or VPN, which emulate circuit behavior within routers. These features mitigate, to some extent, the IP network deficiency for high-quality traffic. It is well known, however, that they tend to degrade the performance of routers, which are most efficient for routing packets that do not contain markers.
A media gateway function is required between an IP network and a TDM network in order to provide voice telephony service between a user connected to the IP network and another user connected to the TDM network. The media gateway transcodes between packet based voice, VoIP, onto a TDM network.
In addition to the media gateway function, it is also possible to incorporate packet switching fabric for VoIP services in the same system. The two switching fabrics may be implemented in a single hardware or in separate hardware but under a common control and signaling infrastructure. A set of signaling and routing protocols for call control such as SS7, SIP, MGCP and others can unify services in an integrated network for both traditional and IP based access means such as voice access over wire pairs and VoIP access such as cable modem and DSL. While the integrated media gateway and circuit-switching system technology offers a means for interworking between the TDM voice network and IP network, voice calls in the IP network are still routed hop-by-hop and packet-by-packet in the IP network.
The telephony network meets the stringent reliability and latency requirements using circuit-switching technology (also known as TDM technology), while an IP network provides flexible and low-cost data communications that do not impose stringent requirements for voice telephony. However, operations and maintenance of a separate voice network for a small fraction of the total traffic is expensive. While an integrated media gateway and voice-switch allows voice calls to flow seamlessly from IP network to TDM network, the cost and complexity to route both high-quality and best-effort traffic in the IP network still remain significant issues. Furthermore, efficient interconnection of voice-switches continues to be problematic. In a network, each switch needs to communicate with every other switch for call set up as well as call routing. However, it is impractical to interconnect every switch with every other switch since such interconnection would require that the number of links grow as the square of the number of switches in a network.
A traditional voice network typically deploys tandem switches, which are intermediary switches that facilitate the interconnection of voice switches. For example, a tandem switch is used to route a call between two voice-switches to which it is connected when the two voice-switches do not have a direct link between them. The tandem switch routs the call in response to a request from one of the two voice-switches. Several layers of tandem switches typically exist in a traditional voice network for providing such connectivity.
An IP network on the other hand is a flat network that routes traffic hop-by-hop. For example, traffic between two routers A and D that are not directly connected but connected through a chain of routers B and C, with links AB, BC and CD, would flow from A to B over link AB, B to C over link BC and finally from C to D over link CD. At each intermediate node B and C the transport links are terminated and all packets are extracted at the intermediate nodes and reassembled in the next link with other traffic. The termination of the transport links and routing via the packet-switching fabrics in routers at each intermediate node is expensive and it is difficult to build larger packet fabric to route transit traffic. In addition, such hop-by-hop transfer adds unnecessary delay in each intermediate node. Thus there is a need for a device to build an integrated network that would meet the same robustness and delay requirements for high-quality traffic while providing flexible, low-cost and scalable transport of best-effort traffic.
FIG. 1 depicts an IP network according to the prior art. IP network 100 is a packet-switched network, i.e. information is carried from one network element to another by means of breaking messages up into discrete, variable length packets. Each packet contains a header section, which includes information about the destination address, source address, packet's priority, etc., and a payload section, which contains the data that makes up a portion of the message. IP network 100 comprises IP routers, 1021 through 1025, which are interconnected in a network configuration via links 1041 through 1047. For example, IP router 1021 is connected to IP routers 1022, 1023, and 1025, via links 1041, 1044, and 1046, respectively. Each of links 1041 through 1047 may be an aggregate link of multiple OC-48 or OC-192 links.
IP router 102i, where i=1 through 5, provides connectivity between user equipment 1061 through 1063 using packetized data transmissions over network links 1041 through 1047 . IP router 102i comprises logic circuitry, memory, and routing information and protocols that enable IP router 102i to receive an information packet, examine the packet, determine the destination for the packet, and decide on an immediate, potentially intermediate, destination for the packet. IP router 102i then transmits the packet to its immediate destination over either a network link (if the final destination is not connected to IP router 102i) or an access link (if the destination is directly connected to IP router 102i). IP router 102i is described below and with respect to FIG. 2.
Network link 104i comprises a plurality of OC-48 and OC-192 signal lines, which carry high-bandwidth transmissions between IP routers (e.g., network link 1041 provides transmissions between IP router 1021 and 1022). In many instances, IP router 104i will aggregate packets that originated as part of different signals into a single transmission at OC-48 or OC-192 rates.
IP network 100 operates in a best-effort, hop-by-hop manner. For example, a transmission by user equipment 1061, which has an intended destination 1063, is spread over multiple packets of data. Each packet includes header information, which contains the intended address for that packet. IP router 1021 receives each packet from user equipment 1061, and examines the destination header of each packet and decides which router to forward the packet to based on an internal connectivity table (i.e., a routing table). Each packet of the transmission may be sent to a different IP router, which is connected to 1021, and may be joined with other packets to compose an OC-48 or OC-192 signal. Each IP router that receives a packet then goes through the same process, wherein it examines the destination header and decides which IP router to which it is connected should receive the packet (i.e., decides on the next hop). Hop-by-hop routing continues until each packet is received by IP router 1024, which is connected to user equipment 1063. The packets that compose the transmission may arrive at user equipment 1063 in any order and each packet can incur various time delays that affect the total delay in user equipment 1063 receiving the transmission in its entirety.
The simplicity of best-effort, hop-by-hop routing and uniformity of signaling and routing protocols make IP network 100 easy to operate. However, IP network 100 is deficient in delivering high-quality services efficiently, in providing scalability and robustness because each intermediate router must route transit traffic packet by packet.
FIG. 2 depicts a schematic diagram of the salient components of an IP router in accordance with the prior art. Router 1024, which is representative of IP routers 1021 through 1025, comprises packet-switching fabric 2104, processor 2124, and link interfaces 222. Processor 2124 comprises logic circuitry and memory and includes routing protocol 2144, link state database 2164, routing table 2184, label switching path database 2204, and signaling protocol 2184.
Packet switching fabric 2104 is a matrix of electronic switches and logic circuitry that receives a packet, reads the destination header of the packet, compares it to the closest match in routing table 2184 to determine the next hop destination, and sends the packet out on the appropriate network link where a link interface assembles the packet into a signal such as a SONET signal ready for transport to the next IP router.
Signaling protocol unit 2184 sends and receives protocol messages through the packet fabric using the links 1043, 1045, and 1047. Based on these messages each router learns the existence of other routers connected in the network and how they are connected i.e., the network topology. As a result, each router creates and maintains the topology information in link state database 2164. Link state database 2164 is used to generate routing table 2184 using a variety of algorithms such as shortest path routing. In addition, if the router has the capability to route a set of packets grouped in a data stream called a Forwarding Equivalency Class (FEC) based on certain criteria such as a higher QoS requirement using labels instead of destination headers. Then IP router 1024 uses signaling protocol such as RSVP and CR-LDP to create label switching paths (LSP) database 2204. LSP database 2204 is then used to route packets based on the label header instead of the destination header of the packet.
FIG. 3 depicts a schematic diagram of the salient components of a voice network in accordance with the prior art. Voice network 300 comprises voice-switches 3121 through 3125 which are connected via cross-connects 3101 through 3105. Cross-connects 3101 through 3105 are interconnected by network links 3041 through 3047. Voice-switches 3121 through 3125 are connected to cross-connects 3101 through 3105 by trunk links 318.
Cross-connect 310i, where i=1 through 5, is a circuit-switching fabric and an associated fabric controller that interconnects any one of N inputs to any one of M outputs. Cross-connect 310i will be described below and with respect to FIG. 4.
Voice-switch 312i, where i=1 through 5, is a DSO-signal-level voice-switching fabric which provides connections between voice terminals 3061 and 3062. The connections between user terminals 3061 and 3062 are made via voice-switches using signaling such as SS7 signaling. Cross-connect 310i provides multiplexing of low-speed signals from the voice-switches to high-speed signals such as OC-48 and OC-192 for transmission to other nodes; provides demultiplexing of a high-speed signal such as OC-48 and OC-192 arriving from a first node; connects one or more low-speed signals from the arriving high-speed signal to the voice-switch connected to cross-connect 310i; and provides multiplexing the remaining low-speed signals within the arriving high-speed signal and one or more low-speed signals arriving from the voice-switch 312i into a high-speed signal such as OC-48 and OC-192 for transmission to a second node in the network.
Cross-connect 310i is typically operated via external commands from a Network Management System (not shown). Based on forecasted demands and traffic patterns a network design is developed which provides the number of low-speed links 318 connecting voice-switches 312i to cross-connect 310i. Based on this connectivity design, the cross-connect management systems configure cross-connects 3101 through 3105 to implement the network design (i.e., the switch connectivity). The connectivity typically remains static for several months until a new network design based on new demand forecast is implemented.
Cross-connect 310i also provides another key transport function, namely, restoration from catastrophic network failures such as a fiber cut. When a failure is detected, cross-connects 3101 through 3105 autonomously detect the failure and reconnect the failed links via an alternate route using network links reserved for restoration.
FIG. 4 depicts a schematic diagram of the salient components of a cross-connect in accordance with the prior art. Cross-connect 3102, which is representative of each of cross-connects 3101 through 3105, comprises circuit-switching fabric 4142, fabric controller 4162, and TDM interfaces 4181 through 4184.
Circuit-switching fabric 4142 is a matrix of electronic logic that interconnects any one of N inputs to any one of M outputs.
Fabric controller 4162 is a processor, which provides control over the connectivity between each of the N inputs and each of the M outputs of circuit-switching fabric 4142. Fabric controller 4162 interprets instructions provided by the Network Management System and translates these instructions into the specific switch configuration required to establish appropriate connectivity within circuit-switching fabric 4142.
TDM interfaces 4181 through 4183 terminate overheads of incoming signals on network links 3041, 3042, and 3043. Overheads include section overheads and line overheads of OC-48 and OC-192 signals arriving at TDM interfaces 4181 through 4183, which are first terminated at the interfaces for network performance monitoring, multiplexing lower level digital bit streams into higher level signals such as OC-48, demultiplexing of higher level signals coming from the network links into lower level digital signals or tributaries. Lower level digital signals in appropriate formats from the TDM interfaces 4181 through 4183 are then sent to circuit-switching fabric 4142. Cross-connect 3102 does not terminate SONET Path overhead since packets inside the SONET payloads are not extracted in the cross-connect 3102.
Circuit-switching fabric 4142 then connects the lower level digital signals (i.e., tributaries) to appropriate interfaces for multiplexing and transporting to the next node. Upon receiving a command from a network management system or detecting a network failure, affected circuit-switching fabrics of cross-connects 3101 through 3105 change their corresponding circuit-switching fabric 4141 through 4145 configurations to, in the case of a received command, provide the connectivity specified by the network management system, or, in the case of a detected network failure, a pre-configured protection circuit. Cross-connects 3101 through 3105 do not participate in signaling that voice-switch 312i uses to set up and control voice calls.